Digital filters are often used to separate or subdivide a continuous time signal into frequency determined portions for subsequent analysis or signal processing. In order to pass an analog signal through a digital filter, the analog signal must first be sampled and then digitized.
Conventional techniques use sampling frequencies sufficiently above the highest frequency of interest so that no significant information is lost in the sampling process. Generally, one tries to use the highest possible sampling frequency and the shortest possible sampling interval in order to most accurately represent the analog signal. However, the sampling frequency is limited by the characteristics of the electronic circuitry and futhermore, the sampling frequency is often a function of cost.
It is well known that the sampling of a continuous time signal produces the phenomenon of frequency aliasing into the signal's frequency spectrum. Frequency aliasing divides the frequency spectrum into sections which are overlaid, thereby in effect folding the frequency spectrum onto itself. Frequencies that are offset by integer multiples of the sampling frequency are overlaid upon each other. This often causes interference, in the form of noise and/or distortion which is overlaid onto the desired signal.